博碩士論文 89521072 詳細資訊




以作者查詢圖書館館藏 以作者查詢臺灣博碩士 以作者查詢全國書目 勘誤回報 、線上人數:52 、訪客IP:3.147.237.38
姓名 廖家慶(Chia-Ching Liau)  查詢紙本館藏   畢業系所 電機工程學系
論文名稱 語者調適之應用研究
(The Research of Speaker Adaptation)
相關論文
★ 小型化 GSM/GPRS 行動通訊模組之研究★ 語者辨識之研究
★ 應用投影法作受擾動奇異系統之強健性分析★ 利用支撐向量機模型改善對立假設特徵函數之語者確認研究
★ 結合高斯混合超級向量與微分核函數之 語者確認研究★ 敏捷移動粒子群最佳化方法
★ 改良式粒子群方法之無失真影像預測編碼應用★ 粒子群演算法應用於語者模型訓練與調適之研究
★ 粒子群演算法之語者確認系統★ 改良式梅爾倒頻譜係數混合多種語音特徵之研究
★ 利用語者特定背景模型之語者確認系統★ 智慧型遠端監控系統
★ 正向系統輸出回授之穩定度分析與控制器設計★ 混合式區間搜索粒子群演算法
★ 基於深度神經網路的手勢辨識研究★ 人體姿勢矯正項鍊配載影像辨識自動校準及手機接收警告系統
檔案 [Endnote RIS 格式]    [Bibtex 格式]    [相關文章]   [文章引用]   [完整記錄]   [館藏目錄]   [檢視]  [下載]
  1. 本電子論文使用權限為同意立即開放。
  2. 已達開放權限電子全文僅授權使用者為學術研究之目的,進行個人非營利性質之檢索、閱讀、列印。
  3. 請遵守中華民國著作權法之相關規定,切勿任意重製、散佈、改作、轉貼、播送,以免觸法。

摘要(中) 摘 要
在語音辨識系統中,特定語者(Speaker-dependent)型語音辨識系統雖有高辨識率的優點,但當應用到新語者時須花釵h語音訓練資料和時間;而不限語者(Speaker-independent)或多語者(multi-speaker)型的語音辨識系統,除最初建立系統時所需語音資料外,應用於新語者時不再需新語音訓練資料,但其辨識率普遍不高。語者調適(Speaker-adaptive)辨識系統則利用一充分訓練過的參考系統已知資訊,藉新語者少量語音資料訓練 ,可達到接近特定語者系統的辨識率,因此論文中將針對語者調適系統進行研究。
本論文內容包含兩個主要研究主軸,其一為如何在少量調適語料之狀況下,增進改善調適演算法,藉此提升系統辨識率與調適結果;另一主軸則為利用增進後之調適演算法實際應用於線上辨識與調適。
於第一研究主軸中,其重點在於考慮初始模型與最大可能性線性迴歸(Maximum Likelihood Linear Regression,MLLR)兩者間貢獻的比重分配,藉由找出最佳平衡點來提升調適性能。接著並考慮向量場平滑化(Vector-Field-Smoothing,VFS)轉移向量場的調適方式,針對沒有觀測到之調適語料模型,加以參考有調適語料之模型來進行調整,藉此特性再搭配權重化之MLLR調適方法研究其調適效果。接者利用特定語者模型與不特定語者模型來架構出特徵向量空間,由此特徵向量空間來找出語者的代表點所在,藉此調整系統模型參數。而在第二研究主軸內,藉由所發展出少量調適語料即能達到調適系統之演算法,將此調適演算法應用於線上系統,使語者能夠感受到辨識與調適之即時變化。
摘要(英) Speaker adaptation has been applied to speech recognition to get a speaker dependent system with a good performance. Most adaptation techniques use the initial model as a starting point and then introduce speaker’s specific information. By using the adapted parameters, the recognition performance can be significantly improved.
In this thesis, we present a variation on improving the performance of maximum likelihood linear regression (MLLR) in cases of little adaptation data. The transformed Gaussian means are interpolated with the means in the initial mean models. The VFS algorithm proposed by the following steps. First, the transfer vectors are estimated. Then, interpolation and smoothing are performed using the transfer vectors. We applied the idea of using eigenvoices, a set of orthogonal basis vectors derived from the parameters of speaker dapendent models trained on reference speakers.
關鍵字(中) ★ 語者調適 關鍵字(英) ★ speaker adaptation
論文目次 目 錄
摘 要…………………………………………………………i
附圖目錄……………………………………………………iv
附表目錄………………………………………………………v
第一章 序論…………………………………………………1
1.1語音處理發展與應用…………………………………………1
1.2語者調適簡述…………………………………………………2
1.3研究動機………………………………………………………3
1.4研究目標………………………………………………………4
1.5論文大綱………………………………………………………5
第二章 語音處理基本技術…………………………………6
2.1特徵參數求取…………………………………………………6
2.2隱藏式馬可夫模型……………………………………………9
第三章 語者調適相關技術…………………………………16
3.1 修正最大可能性線性迴歸…………………………………16
3.1.1最大可能性線性迴歸理論……………………………16
3.1.2 權重化MLLR調適方法之推導………………………17
3.1.3最大可能性線性迴歸其對角化 之推導……………20
3.2向量場平滑化(VFS) ………………………………………24
3.3權重化MLLR調適方法與向量場平滑化之合併應用………26
3.4特徵語音調適法(Eigenvoice) ……………………………27
第四章 系統模型架構………………………………………29
4.1 系統次音節模型架構…………………………………29
4.2 系統次音節模型訓練與辨識………………………………32
第五章 研究方法與結果…………………………………………36
5.1研究環境………………………………………………………36
5.1.1系統特徵參數與研究設備……………………………36
5.1.2訓練、調適及測試語料…………………………………36
5.2權重化修正MLLR調適之研究………………………………37
5.3權重化修正MLLR與VFS結合調適之研究………………40
5.4特徵語音調適法之研究………………………………………42
5.5辨識與調適應用系統…………………………………………44
第六章 結論與展望…………………………………………47
6.1結論……………………………………………………………47
6.2展望……………………………………………………………48
參考文獻……………………………………………………49
參考文獻 參考文獻
[1] Rabiner,L. R. et al.”Recognition of Isolated Digits Using Hidden
Markov Models with Continuous Mixture Densities,” AT&T Technical Journal 64(6):1211-1233,1985.
[2] Juang,B.H.,and Rabiner,L.R.”Mixture Auto-regressive Hidden
Markov Models for Speech Signals,”IEEE Trans.on ASSP,vol.33,No.6,pp.1404-1413,Dec.1985.
[3] Rabiner,L.R.,and Juang,B.H. ”An Introduction to Hidden Markov
Models”IEEE ASSP Magzine,Jan.1986.
[4] L.R. Rabiner, “A Tutorial on Hidden Markov Models and Selected Applications in Speech Recognition”. Proc. IEEE, Vol. 77, No.2, pp. 257–286, Feb. 1989.
[5] C-H. Lee, C-H. Lin, and B-H. Juang, “A Study on Speaker Adaptation of the Parameters of Continuous Density Hidden Markov Models”. IEEE Trans. on Sig. Proc., Vol. 39, No. 4, pp. 806–814, April 1991.
[6] Heidi Christensen, “Speaker Adaptation of Hidden Markov Models using Maximum Likelihood Linear Regression”. MSc.E.E. Thesis. Aalborg University, Denmark, June 1996.
[7] C.J. Leggetter and P.C. Woodland, “Speaker Adaptation of HMM’s using Linear Regression”. Technical Report GUED/F-INFENG/ TR.181, Cambridge University, June 1994.
[8] C.J. Leggetter and P.C. Woodland, “Maximum Likelihood Linear Regression for Speaker Adaptation of Continuous Density Hidden Markov Models”. Computer Speech and Language, Vol. 9, pp. 171–185, 1995.
[9] C.J. Leggetter and P.C. Woodland, “Flexible Speaker Adaptation using Maximum Likelihood Linear Regression”. Proc. ARPA Spoken Language Technology Workshop, pp. 104–109, Feb. 1995.
[10] C.J. Leggetter and P.C. Woodland, “Speaker Adaptation of continuous density HMMs using Multivariate Linear Regression”. ICSLP-94, Vol. 2, pp. 451–454, Yokohama, 1994.
[11] M.J.F. Gales, “Maximum Likelihood Linear Transformation for HMM-Based Speech Recognition”. Technical Report GUED/F-INFENG/TR.291, Cambridge University, May 1997.
[12] M.J.F. Gales, “The Generation and use of Regression Class Trees for MLLR Adaptation”. Technical Report GUED/F-INFENG/TR.263, Cambridge University, August 1996.
[13] A. Sankar and C-H. Lee, “A Maximum-Likelihood Approach to Stochastic Matching for Robust Speech Recognition”. IEEE Trans. on Speech and Audio Proc., Vol. 4, pp. 190–202, May 1996
[14] L. R. Rabiner and R. W. Schafer, “ Digital Processing of Speech Recognition Signals ”, Prentice-Hall Co. Ltd, 1978.
[15] M. Tonomura, T. Kosaka and S. Matsunaga, “Speaker Adaptation Based on Transfer Vector Field Smoothing using Maximum a Posteriori Probability Estimation”. ICASSP-95, Vol. 1, pp. 688–691, 1995.
[16] B.F. Necioglu, M. Ostendorf, and J.R. Rohlicek, “A Bayesian Approach to Speaker Adaptation for the Stochastic Segment Model”. ICASSP-92, Vol. 1, pp. 437–440, 1992.
[17] J-I. Takahashi and S. Sagayama, “Fast Telephone Channel Adaptation Based on Vector Field Smoothing Technique”. Second IEEE Workshop on Interactive Voice Technology for Telecommunications Applications, pp. 97–100, 1994.
[18] J. Takahashi and S. Sagayama, “Vector-Field-Smoothed Bayesian Learning for Incremental Speaker Adaptation”. ICASSP-95, Vol. 1, pp. 696–699, 1995.
[19] J. Takahashi and S. Sagayama, “Minimum Classification Error Training for a Small Amount of Data Enhanced by Vector-Field-Smoothed Bayesian Learning”. ICASSP-96, Vol.: 2, pp. 597–600, 1996.
[20] R. Kuhn, P. Nguyen, J. –C. Junqua, N. Niedzielski, “Rapid Speaker Adaptation in Eigenvoice Space”. IEEE Trans. on Speech and Audio Proc., Vol. 8, pp. 695-707, Nov. 2000.
指導教授 莊堯棠(Yau-Tarng Juang) 審核日期 2002-6-6
推文 facebook   plurk   twitter   funp   google   live   udn   HD   myshare   reddit   netvibes   friend   youpush   delicious   baidu   
網路書籤 Google bookmarks   del.icio.us   hemidemi   myshare   

若有論文相關問題,請聯絡國立中央大學圖書館推廣服務組 TEL:(03)422-7151轉57407,或E-mail聯絡  - 隱私權政策聲明